Voice over IP (“VoIP”) is a relatively recent development that is utilized to transmit voice conversations over a data network using the Internet Protocol (“IP”). Internet Protocol is a part of the TCP/IP family of protocols described in software that tracks the Internet address of nodes, routes outgoing messages, and recognizes incoming messages. Such a data network may be the Internet or a corporate intranet, or any TCP/IP network. There are several potential benefits for moving voice over a data network using IP. First, there is a savings in money compared to the need to use traditional tolled telecommunications networks. Additionally, Voice over IP enables the management of voice and data over a single network. And, with the use of IP phones, moves, adds and changes are easier and less expensive to implement. Moreover, additional and integrated new services, including integrated messaging, bandwidth on demand, voice e-mails, the development of “voice portals” on the Web, simplified setting up and tearing down, and transferring of phone calls are capable.
Using Voice over IP technology, phone systems can communicate with each other over existing TCP/IP data networks typically present between remote offices. This feature alone can eliminate the need for expensive, dedicated circuits between facilities. The shared bandwidth can also be used for voice calls and data communication simultaneously; no bandwidth is dedicated to one or the other.
Another advantage of a Voice over IP system is the ability to implement a phone system over an existing data network that is already connecting workstations within a local area network (LAN) and even over a wide area network (WAN). Such networks utilize frame packets for transmitting information. Voice over IP can utilize such packet switching capabilities to connect IP phones onto the LAN and/or WAN. However, the implementation of Voice over IP onto a network has proven to have some difficulties. Data networks were originally designed to allow for latency (delays) in the delivery of packets between sources and destinations. If a packet became lost, then the network would go through a re-send protocol to have the packet sent again from the source to the destination, and the data then reassembled at the destination end. With voice (or for that matter, video or any other real-time application), such delays present problems. Real-time applications cannot tolerate significant delays or they no longer become real-time applications. Such quality of service (“QOS”) concerns are especially amplified when attempting to implement Voice over IP onto a data network, since it can be affected by bursts of data transfers among the workstations and servers, etc. For example, a large print job or a file access can significantly occupy the bandwidth on such a network, thus greatly degrading the ability to transmit any real-time information during that data burst. This problem worsens as more and more Voice over IP telephones are added to the network.
FIG. 1 illustrates a typical configuration where a remote IP telephone is connected to an isolated network, consisting of a NAT (Network Address Translation) router, the IP telephone, and a personal computer (PC). The other side of the NAT router is typically connected to a WAN such as xDSL, cable or ISDN, through the use of a modem. The data bandwidth of a WAN connection is usually very limited compared to that of a LAN. This limited bandwidth is shared among all devices connected to the NAT router. A problem arises when the remote IP telephone is in use, and the workstation (PC) attempts to access a large amount of data. During this condition, it is possible for the workstation to essentially “starve” the IP telephone of bandwidth, causing the IP telephone to malfunction (e.g., broken speech, etc.). Because it is preferred that the IP telephone communicate its content uninterrupted, there is a need in the art to provide the IP telephone with guaranteed priority over workstation data.